Difference between revisions of "Collaboration with VoIP Providers"
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*First the VOIP / SIP trunking providers, which are most important for your business, are selected and agreed with [mailto:franz.kneissl@unify.com product management]. | *First the VOIP / SIP trunking providers, which are most important for your business, are selected and agreed with [mailto:franz.kneissl@unify.com product management]. | ||
| − | *Next step is to get the IP and SIP settings necessary for interworking with the specific provider. The document "VoIP Provider | + | *Next step is to get the IP and SIP settings necessary for interworking with the specific provider. The document "VoIP Provider Data Collection" has to be completed and sent to the [mailto:anastasios.ginis@unify.com;karl-werner.weigt@unify.com VoIP Provider cerification team] to allow correct initial PBX setup. |
*After that, the VoIP provider test list has to be worked through in the country, according to the instructions included. Remote support is available by the [mailto:anastasios.ginis@unify.com;karl-werner.weigt@unify.com VoIP Provider cerification team]. The tests cover the most important features at the SIP trunking interface. Leading switch should be the OpenScape Office MX, after that a reduced delta test list can be used for HiPath 3000. | *After that, the VoIP provider test list has to be worked through in the country, according to the instructions included. Remote support is available by the [mailto:anastasios.ginis@unify.com;karl-werner.weigt@unify.com VoIP Provider cerification team]. The tests cover the most important features at the SIP trunking interface. Leading switch should be the OpenScape Office MX, after that a reduced delta test list can be used for HiPath 3000. | ||
| − | {{File-DL|VOIP provider test list and configuration|zip}} last update: 2011- | + | {{File-DL|VOIP provider test list and configuration|zip}} last update: 2011-07-07 |
*The final test results including traces and provider data shall be delivered to the [mailto:anastasios.ginis@unify.com;karl-werner.weigt@unify.com VoIP Provider cerification team]. | *The final test results including traces and provider data shall be delivered to the [mailto:anastasios.ginis@unify.com;karl-werner.weigt@unify.com VoIP Provider cerification team]. | ||
*The appropriate VoIP provider configuration data will be included in the HiPath / OpenScape Office administration for easy setup. | *The appropriate VoIP provider configuration data will be included in the HiPath / OpenScape Office administration for easy setup. | ||
Revision as of 09:08, 7 July 2011
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 Home Communications Systems Collaboration with VoIP Providers
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File:VoIP Provider SME.jpg
VoIP Providers
Collaboration with VoIP Providers (also known as ITSP) via SIP Trunking is an important topic in the todays world of IP Voice/Data Communication. The following information will give an overview of the supported VoIP Providers in the different countries as well as additional considerations and hints regarding the SME platforms.
You did not find your provider? Find out how to get it released!
Contents
- 1 Overview
- 2 General Information by platform
- 3 Tested VoIP Providers by Countries
- 3.1 International
- 3.2 Australia
- 3.3 Austria
- 3.4 Belgium
- 3.5 Brazil
- 3.6 Czech Republic
- 3.7 Chile
- 3.8 Finland
- 3.9 France
- 3.10 Germany
- 3.11 Italy
- 3.12 Mexico
- 3.13 The Netherlands
- 3.14 Portugal
- 3.15 Slovenia
- 3.16 South Africa
- 3.17 Spain
- 3.18 Sweden
- 3.19 Switzerland
- 3.20 United Kingdom
- 3.21 United States of America
- 4 Released SIP Providers in Detail
- 5 Frequently asked Questions
- 6 External Links
Overview
The SME platforms have been tested with a large number of Internet telephony providers that support SIP. Continuously new providers are tested, released and included within administration. As the SIP recommendations leave some room for interpretation, there are differences in the range of features supported with a specific provider. In this chapter general information is given which is valid for all platforms and all SIP Providers.
SIP trunking features for all SME platforms
- up to 4 active SIP providers and one Internet connection
- operation behind another router using STUN
- SIP trunking with single numbers (MSN) and up to 30 SIP user accounts (called Internet telephony stations in WBM)
- Direct Inward Dialing (DID)
- most important features available
- CLIP
- Hold, Consultation, Toggle
- Transfer (attended, semi-attended)
- Ringing group, Call pickup
- Conference
- Call Forwarding
- DTMF transmission e.g. for voicemail access
- large number of simultaneous VoIP connections depending on available DSL bandwidth, used Codec and system. G.711 needs up to 92 kbit/s, G.729A needs up to 36 kbit/s in both directions.
- automatic fallback to ISDN at SIP trunk failure
- Voice Codecs G.711 and G.729
- Fax over IP via T.38 (or G.711)
| Security Warning - Malicious attacks from the Internet may lead to reduced service quality or toll fraud! Please strictly follow these basic rules:
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The SME platforms with SIP Stack 4.0.26 have been validated with the BroadWorks SIP Interface. BroadWorks is the Carrier VoIP solution from Broadsoft, the global market leader. The tested SIP Stack is used in the current system versions Hipath 3000 V8 and OpenScape Office V3 as well.
Broadsoft Validation - Broadsoft Partner Configuration Guide (see configuration guides for the current systems below)
General Requirements for VoIP
- LAN with 10/100/1000 MBit/s and no more than 40% network load
- Separate port on the switch or router for every component in the IP network (no hubs as concentrators)
- Not more than 50 ms delay in one direction (One Way Delay); not more than 150 ms total delay
- Not more than 3% packet loss and 20 ms jitter
- Quality of Service (QoS) support - IEEE 802.p, DiffServ (RFC 2474) or ToS (RFC 791)
- Sufficient WAN bandwidth (uplink and downlink) for intended simultaneous calls and CODEC
- External modem, such as a DSL modem
- Intermediate routers shall support quality of service and bandwidth control mechanisms if the WAN link is used for voice and data simultaneously. SIP messages have to be routed transparently.
For interworking with different network configurations and edge devices and how to use STUN please see Network Configuration for VoIP Providers. One common STUN setting must be used for all providers in a system.
Restrictions for all Providers and SME Platforms
- Special call numbers and emergency calls
- Special call numbers (e.g. 0800, 0900, 0137) or analog fax/modem connections are not supported by all SIP providers.
- Emergency call numbers (e.g. 110, 112) are not supported by most providers. These connections must be implemented via S0 access (administration default)
- Only one external destination is allowed for ringing groups which are to be called via SIP trunk from VoIP provider.
- DTMF is supported via RFC 2833 or inband. SIP INFO method is not supported.
- For Fax transmission the T.38 protocol is highly recommended. Fax pass through using codec G.711 has been found much less reliable in most SIP provider environments. If your provider does not yet support Fax T.38, Fax pass through with G.711 is possible but there is no warranty or further support.
- Other media types besides audio (voice and voice band data) and image (fax) are not supported.
- The ITSP has to support outband DTMF (RFC2833) for using the OpenScape Office Autoattendant via SIP trunk.
- CLIP no screening is not available with our platforms for SIP trunks. Please check with your ITSP, if this can be accomplished at the provider side.
How to get a new VoIP provider released
Tests are necessary with every new provider before making installations in the field due to different implementations of the SIP protocol.
- First the VOIP / SIP trunking providers, which are most important for your business, are selected and agreed with product management.
- Next step is to get the IP and SIP settings necessary for interworking with the specific provider. The document "VoIP Provider Data Collection" has to be completed and sent to the VoIP Provider cerification team to allow correct initial PBX setup.
- After that, the VoIP provider test list has to be worked through in the country, according to the instructions included. Remote support is available by the VoIP Provider cerification team. The tests cover the most important features at the SIP trunking interface. Leading switch should be the OpenScape Office MX, after that a reduced delta test list can be used for HiPath 3000.
VOIP provider test list and configuration last update: 2011-07-07
- The final test results including traces and provider data shall be delivered to the VoIP Provider cerification team.
- The appropriate VoIP provider configuration data will be included in the HiPath / OpenScape Office administration for easy setup.
- After successful tests the VoIP provider is released, and published on this page. Back level support is available for the new provider after that.
General Information by platform
For discontinued platforms see Collaboration with VoIP Providers (discontinued platforms).
HiPath 3000
Valid for HiPath 3000 ≥ V7 with HG 1500:
SIP Trunking features
- For common features and restrictions see overview
- An external router is always necessary
- Maximum 32 concurrent calls between IP phones and ITSP (depending on the available bandwidth and used Codec). For TDM devices (analog, ISDN) 1 B-channel at HG 1500 has to be available for each concurrent call (gateway call).
- SIP trunks can not be used for dialing through or direct inward system access with DTMF tones e.g for the feature Mobility Entry. Please use a TDM line for those features.
- No FAX via ITSP from/to OpenScape Office HX client applications. A TDM trunk has to be used.
- A trunk prefix to select a SIP trunk as an alternate trunk can be used from the OpenScape Office HX client's dial box but not from client's directories and settings (myPortal, myAttendant, myAgent).
- SIP trunking is not released for use with the OpenScape Office HX Contact center application (planned for HiPath 3000 V9).
Configuration
HiPath 3000 Configuration Guide for Internet Telephony:
See also: How to enter SIP Provider Account Data
OpenScape Office LX/MX
Valid for OpenScape Office V2 (MX) and V3 (LX/MX):
SIP Trunking features
- All features, requirements and restrictions listed in overview are valid.
- Up to 32 simultaneous calls via SIP trunking for OpenScape Office MX (depending on available DSL bandwidth and used Codec)
- Up to 128 simultaneous calls via SIP trunking for OpenScape Office LX (depending on available DSL bandwidth and used Codec)
- An external router is always necessary for OpenScape Office LX, OpenScape Office MX can optionally be operated with the bulit-in router / firewall.
- SIP trunks cannot be used for dialing through or direct inward system access with DTMF tones e.g. for the feature Mobility Entry. Please use a TDM line for those features. Starting from OpenScape Office V3, myPortal for Mobile supports comprehensive control and UC functionality.
- FAX over IP with T.38 is supported for analog Fax devices, terminal adapters and the OpenScape Office application. Fax via G.711 is not supported with the application. No Fax over IP is available for the OpenScape Office V2 application, a TDM Trunk has to be used.
- A trunk prefix to select a SIP trunk as an alternate trunk can be used from the OpenScape Office client's dial box but not from client's directories and settings (myPortal, myAttendant, myAgent).
- SIP trunks cannot be used for presence based call forwarding, if a TDM trunk is also present.
- Release of SIP trunks for use with the OpenScape Office Contact Center application is under preparation for OpenScape Office V3.
Configuration
OpenScape Office Configuration Guide for Internet Telephony:
See also: How to enter SIP Provider Account Data
Tested VoIP Providers by Countries
The main SIP trunking functions of the SME platforms have been tested with the following Internet telephony providers. For detailed information see the document Released SIP Providers below.
= has been successfully tested by development or customer
DID = accounts for PBX-trunking / Direct Inward Dialling
MSN = accounts for single number(s)
International
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| COLT | DID |
|
|
| Skype Connect | MSN |
|
|
- Please check provider home page for supported countries.
- Skype Connect (former name Skype for SIP) is a fee-based business service from Skype. It allows voice communication between Skype clients and office phones using the standard SIP trunking interface. Other Skype client features like chat or video are currently not supported.
Our SME Platforms have been officially certified by Skype. Skype Connect with SME platforms
Australia
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Engin | MSN |
|
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| Internode | MSN |
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|
Austria
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| UPC / inode | MSN |
|
- |
| IP-austria | DID |
|
|
| WNT | DID |
|
- |
Belgium
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Belgacom | DID |
|
|
| Hexacom | MSN DID |
|
- |
| Telenet | MSN DID |
|
|
Brazil
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| CTBC 1) | DID |
|
- |
| GVT 1) | DID |
|
|
Czech Republic
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| T-mobile | MSN DID |
|
|
| Telefonica O2 | MSN DID |
|
|
Chile
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Entel NGN | MSN |
|
- |
Finland
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Elisa | DID |
|
- |
| Telia-Sonera | DID |
|
|
France
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| OpenIP | DID | - |
|
| Optimitel | DID |
|
- |
Germany
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| 1&1 | MSN |
|
- |
| Vodafone (formerly Arcor) | VoIP Anlagenanschluss (DID) |
|
|
| freenet (now 1&1) | MSN |
|
- |
| freenetBusiness (discontinued) | DID |
|
|
| GMX (now 1&1) | MSN |
|
- |
| Purtel | MSN |
|
- |
| QSC | IPfonie extended (DID) |
|
|
| sipgate | sipgate basic (MSN) sipgate trunking (DID) |
|
|
| toplink | MSN DID |
|
|
| XSip (discontinued) | DID |
|
- |
Italy
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Infracom (ex Multilink) | MSN |
|
- |
| TWT | MSN DID |
| |
| Viatek | MSN DID |
|
|
Mexico
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Telefonica Mexico 1) | MSN | - |
|
The Netherlands
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| I2Connect | MSN DID |
|
- |
| Infopact | MSN DID |
|
- |
| OneCentral | MSN DID |
|
|
| Priority Telecom | MSN |
|
|
| Tele2 | MSN DID |
- | 1)1)
|
1) Only for OpenScape Office MX. The WAN interface with PPPoE has to be used, no operation behind an external router is supported.
Portugal
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Portugal Telecom | DID |
|
|
Slovenia
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Amis | MSN |
|
|
| Detel | MSN |
|
|
| MegaM | MSN |
|
|
| tustelekom | DID |
|
|
| T-2 | DID | - |
|
| UPC Telemach | MSN |
|
|
South Africa
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Telfree | MSN DID |
- |
|
| Vodacom | MSN DID |
|
|
Spain
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| vozelia | MSN DID |
|
- |
| Xtra Telecom | MSN |
|
|
Sweden
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Tele2 | DID |
|
|
| Telia | MSN |
|
|
Switzerland
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Cablecom | DID |
|
|
| sipcall | MSN | - |
|
| SofiCall | MSN | - |
|
| vtx | DID |
|
|
United Kingdom
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Gamma | DID |
|
|
| Opal | DID |
|
|
| Voiceflex | DID |
|
|
United States of America
| VoIP Providers | Type | HiPath 3000 | OpenScape Office |
|---|---|---|---|
| Cbeyond | MSN |
|
- |
| Sotel | MSN DID |
- |
|
| XO 1) | MSN DID |
- |
|
1) Access parameters depend on the region within the country. If no default provider profile is supplied within administration, please use "create profile" for configuration.
Released SIP Providers in Detail
The table provided here shows the ITSP providers that were successfully checked in a connectivity test. More details about test results are given in the pdf-file Overview of released SIP providers below. Informations concerning CLIP, CLIR, COLP etc. are listed, restrictions may be reported as well.
- Overview of released SIP providers Latest update: 2011-05-12
Connectivity Test Information
The connectivity test is carried out with
- HiPath 3000
- OpenScape Office MX/LX
Tests are done for MSN- and DID-account (as long as the provider´s realm is different), also the platforms are checked as NAT router and / or behind a router.
Several test scenarios are clustered in the test list, main issues are:
- Registration
- Basic Calls from / to ITSP
- Feature Tests
- Special Tests
- emergency calls
- service calls
- long duration calls
The connectivity tests are performed using the following endpoints:
- Analogue
- ISDN
- HFA-phones
- UP0 phones
- SIP-phones
- AP 1120 HFA / SIP
- optiClient 130 HFA
- WLAN2-phones HFA / SIP
At least 3 different types of phones are used for each test case.
Frequently asked Questions
- VoIP Provider FAQ - answers to frequently asked questions
External Links
- Siemens Enterprise Business Area (SEBA) (login required, Internet Explorer only)
VoIP Anbieter (Übersicht, Feature, Vergleiche)
